Index: webrtc/media/engine/webrtcvoiceengine.h |
diff --git a/webrtc/media/engine/webrtcvoiceengine.h b/webrtc/media/engine/webrtcvoiceengine.h |
index 55655e3887f20f9fa1fc587f9c7795b7fbe0e93a..dbb7ea6e76715a811b75a9b907c0e7c4b077fef1 100644 |
--- a/webrtc/media/engine/webrtcvoiceengine.h |
+++ b/webrtc/media/engine/webrtcvoiceengine.h |
@@ -12,12 +12,12 @@ |
#define WEBRTC_MEDIA_ENGINE_WEBRTCVOICEENGINE_H_ |
#include <map> |
+#include <memory> |
#include <string> |
#include <vector> |
#include "webrtc/audio_state.h" |
#include "webrtc/base/buffer.h" |
-#include "webrtc/base/scoped_ptr.h" |
#include "webrtc/base/stream.h" |
#include "webrtc/base/thread_checker.h" |
#include "webrtc/call.h" |
@@ -113,7 +113,7 @@ class WebRtcVoiceEngine final : public webrtc::TraceCallback { |
rtc::ThreadChecker worker_thread_checker_; |
// The primary instance of WebRtc VoiceEngine. |
- rtc::scoped_ptr<VoEWrapper> voe_wrapper_; |
+ std::unique_ptr<VoEWrapper> voe_wrapper_; |
rtc::scoped_refptr<webrtc::AudioState> audio_state_; |
// The external audio device manager |
webrtc::AudioDeviceModule* adm_ = nullptr; |
@@ -188,7 +188,7 @@ class WebRtcVoiceMediaChannel final : public VoiceMediaChannel, |
void SetRawAudioSink( |
uint32_t ssrc, |
- rtc::scoped_ptr<webrtc::AudioSinkInterface> sink) override; |
+ std::unique_ptr<webrtc::AudioSinkInterface> sink) override; |
// implements Transport interface |
bool SendRtp(const uint8_t* data, |
@@ -240,7 +240,7 @@ class WebRtcVoiceMediaChannel final : public VoiceMediaChannel, |
WebRtcVoiceEngine* const engine_ = nullptr; |
std::vector<AudioCodec> recv_codecs_; |
std::vector<AudioCodec> send_codecs_; |
- rtc::scoped_ptr<webrtc::CodecInst> send_codec_; |
+ std::unique_ptr<webrtc::CodecInst> send_codec_; |
bool send_bitrate_setting_ = false; |
int send_bitrate_bps_ = 0; |
AudioOptions options_; |
@@ -258,7 +258,7 @@ class WebRtcVoiceMediaChannel final : public VoiceMediaChannel, |
// Volume for unsignalled stream, which may be set before the stream exists. |
double default_recv_volume_ = 1.0; |
// Sink for unsignalled stream, which may be set before the stream exists. |
- rtc::scoped_ptr<webrtc::AudioSinkInterface> default_sink_; |
+ std::unique_ptr<webrtc::AudioSinkInterface> default_sink_; |
// Default SSRC to use for RTCP receiver reports in case of no signaled |
// send streams. See: https://code.google.com/p/webrtc/issues/detail?id=4740 |
// and https://code.google.com/p/chromium/issues/detail?id=547661 |